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| 00:10.00 | ChkDigit | Got it. hanguponpolarityswitch=yes |
| 00:25.29 | WIMPy | wonders if notzopsi is somehow related to the famous notmes. |
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| 00:47.07 | TazzNZ | back |
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| 07:51.44 | stevePearPear | hi, could I check what does callbackextension for realtime Asterisk do? Purely numeric hostname (123), and not a peer--rejecting! |
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| 07:59.52 | ChannelZ | Well callbackextension asks the peer to send its calls to a specific extension. It's not a hostname. |
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| 09:33.41 | ruied | Hi, I would like to dial in to asterisk trough gsm/sip, and once asterisk recognizes the calling number, it will let the calling person to make an outside call trough sip. This part is working (I'm using waitexten ), my problem is that I want from android to dial to the pbx with the two numbers like XXXXXXXXXX#YYYYYYYYYYYY X is the dial to asterisk, # is the pause in android and Y is the asterisk outbound sip provider call |
| 09:36.13 | ruied | I want to dial the two numbers in just one call, but asterisk doesn't take the YYYYYYYYY number. I think it might be the pause code, but not sure if it is that or if it could be some error in asterisk... |
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| 10:41.55 | _MoneY_ | Is there a programm to leech events like ringing, busy, not available, answered, etc.? |
| 10:42.32 | _MoneY_ | I dont't like to recompile the source to add some lines. |
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| 10:52.19 | zaibb | hi |
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| 11:15.51 | WIMPy | ruied: Chech the manual of your phone. |
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| 12:37.26 | Varazir | I have installed asteriskNOW and i'm looking for some newbe guides/howto. What I like to do is when someone calls in they should get a" please press 1 for heldesk, press 2 for sale, press 3 to leav msg" and so on, But i'm not sure how to set that up and what to search for. I have trunk set up that's all. |
| 12:41.30 | defswork | do you need reinvite to allow switching to video during a call (using linphone) ? |
| 12:45.26 | wdoekes | ~asterisknow |
| 12:45.26 | infobot | from memory, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 12:46.14 | wdoekes | defswork: switching (adding/removing) codecs will require a reinvite. |
| 12:46.25 | wdoekes | but that shouldn't be a problem |
| 12:46.40 | defswork | I can see the SIP media change |
| 12:47.20 | defswork | cant really see anything rejecting it |
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| 12:55.44 | [TK]D-Fender | Varazir: Applications > IVR |
| 12:55.59 | [TK]D-Fender | Varazir: And Join #freepbx for all support concerning the GUI |
| 12:56.23 | Varazir | [TK]D-Fender: Okay thanks :) |
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| 14:04.16 | wrxed | Not sure who did the awesome update to the AMI specifications, but thank whomever that was. I do have a question though, specifically what is a new state ring vs new state ringing? |
| 14:06.20 | slummer | I'm using asterisk 11.10.0 compiled from source and Digium phones with DPMA. The DPMA module works fine, but it has disabled the MWI on the phones. |
| 14:08.46 | slummer | MWI for voice mail |
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| 14:15.06 | newtonr | wrxed, I think that was mjordan who did the AMI spec update |
| 14:17.53 | wrxed | thanks @newtonr |
| 14:18.28 | Stefan27 | My s300 snom phone registered in asterisk 12 in sip.conf gets error chan_sip.c:10663 process_sdp: We are requesting SRTP for audio, but they responded without it! when it tries to make a call. (But when the same device registers at using chan_pjsip there's no error.) 'sip show peer myphones300' shows Encryption:no so why does asterisk request SRTP? |
| 14:21.03 | newtonr | slummer, call Digium tech support, they can best help you out. |
| 14:21.51 | slummer | newtonr: don't I need a service contract for that? |
| 14:22.23 | [TK]D-Fender | slummer: That's installation support for the phones of theirs you bought... |
| 14:23.29 | newtonr | slummer, no you don't for basic support getting the phones working with asterisk/DPMA |
| 14:23.52 | slummer | k |
| 14:23.55 | slummer | thanks |
| 14:24.51 | mjordan | wrxed: that's a tricky question :-) |
| 14:25.46 | mjordan | wrxed: often it depends on the channel driver in question. Typically - although there's probably a lot of caveats to this statement - RING is that the channel has a ringing tone being played on it (regardless of the channel being inbound/outbound). RINGING is when we detect that the far end point is ringing. |
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| 14:25.58 | mjordan | wrxed: it can, however, depend heavily on the channel driver in question. |
| 14:26.22 | mjordan | wrxed: if you are writing an external application, and you want to know whether or not the channel is in a state where a ringing indication is being heard, I would handle both. |
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| 14:27.43 | mjordan | Stefan27: that error statement in chan_sip doesn't really convey when all that situation occurs. Currently, if something offers SRTP crypto attributes to chan_sip and encryption is set to no, it will reject the call |
| 14:28.02 | mjordan | Stefan27: I would suspect that if you look at the inbound INVITE request, it will show that the Snom offered optional encryption |
| 14:29.11 | wrxed | Thank you MJordan |
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| 15:14.10 | Stefan27 | thank you mjordan ill try dig deeper |
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| 16:02.09 | file | falls into existence |
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| 16:36.11 | qakhan | i am trying to run a php script but it is not working, here is my dial plan and cli |
| 16:36.21 | qakhan | http://pastebin.com/fC4j62BM |
| 16:36.29 | qakhan | http://pastebin.com/YLaUFMXw |
| 16:38.51 | [TK]D-Fender | And the one thing you didn't show is the thing that it is complaining about... |
| 16:39.08 | qakhan | :P |
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| 16:40.26 | qakhan | http://pastebin.com/DNmpgHfs |
| 16:43.07 | [TK]D-Fender | Your PHP is using relative paths.... |
| 16:43.39 | qakhan | you mean script path? |
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| 16:45.44 | [TK]D-Fender | clearly |
| 16:45.59 | qakhan | yes |
| 16:46.25 | qakhan | if i run this script directly it works fine |
| 16:46.45 | qakhan | but not working through AGI |
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| 16:54.29 | [TK]D-Fender | I'm not hearing you changing and testing.... |
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| 16:57.37 | qakhan | what? i did not get you |
| 16:58.25 | [TK]D-Fender | CHANGE THE PATHS. |
| 16:59.25 | qakhan | script path? |
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| 17:03.20 | *** join/#asterisk pjensen00 (~per@ip-64-21-247-189.far.ideaone.net) |
| 17:07.05 | pjensen00 | Hello, I'm looking to add the SIP call ID to an event in asterisk. However I know the events should be channel technology agnostic. I noticed the SIP call ID does get set in the channel variable BRIDGEPVTCALLID upon a successful two caller bridge. I was wondering if there was a chance to get upstream acceptance into Asterisk 12 if I were to patch a more generic "CallID" and "DestCallID" on the "DialBegin" event. |
| 17:09.18 | qakhan | [TK]D-Fender i moved script to other folder, its working now |
| 17:09.40 | qakhan | what is the reasone? |
| 17:09.51 | [TK]D-Fender | [12:58][TK]D-FenderCHANGE THE PATHS. |
| 17:10.34 | qakhan | script path? or php ? |
| 17:10.58 | [TK]D-Fender | [12:43][TK]D-FenderYour PHP is using relative paths.... |
| 17:11.25 | qakhan | ok |
| 17:11.29 | qakhan | thanks |
| 17:14.08 | qakhan | 1 question. how to get veriable value from php script in dial plan |
| 17:14.13 | qakhan | Noop() |
| 17:14.48 | mjordan | pjensen00: Why do you want the Call ID on the DialBegin event? |
| 17:15.21 | [TK]D-Fender | qakhan: You're alrady driving this car as if it were a skateboard... |
| 17:15.51 | [TK]D-Fender | qakhan: You shouldn't have to be passing args to an AGI... it already has access to all the channel variables |
| 17:16.01 | pjensen00 | mjordan: For internal reporting implementation |
| 17:16.03 | [TK]D-Fender | qakhan: And that is what you'd set before leaving |
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| 17:17.10 | pjensen00 | mjordan: We already did a small patch making SIP headers accessible on the outgoing leg of a dial. I am able to get them on a hangup callback, but for reasons of how our reporting is structured I need to be able to report the call ID on DialBegin |
| 17:17.39 | mjordan | pjensen00: With a pre-dial handler, I'd expect you can get the SIP headers (including the Call-ID) on the outbound leg |
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| 17:18.45 | mjordan | pjensen00: in the pre-dial handler, you could set a channel variable to the value of the SIP call ID |
| 17:19.06 | mjordan | pjensen00: then, using the cannelvars setting in manager.conf, you'd get the channel variable on every channel based event, which includes DialBegin |
| 17:19.24 | mjordan | er, channelvars |
| 17:19.26 | pjensen00 | mjordan: the pre-dial handler doesn't have access to the SIP Call ID, To, From or any other SIP header yet. I only am able to access them after the dial has already begun |
| 17:19.32 | mjordan | the pre-dial handlers *should* be executed prior to DialBegin |
| 17:19.40 | mjordan | That is true. Hm. |
| 17:19.47 | pjensen00 | Unless I mis handled the patch and the SIP headers are available earlier |
| 17:20.08 | pjensen00 | I tried using the "on bridge" option on the dial which works.... except whent he call isn't picked up |
| 17:20.09 | mjordan | No, pre-dial is done after channel creation but before any dial operation takes place, which means the INVITE request hasn't be created yet. Nuts. |
| 17:20.19 | mjordan | been created. |
| 17:20.26 | mjordan | Hm. |
| 17:20.29 | pjensen00 | Yeah, that was my "awww fudge" moment |
| 17:20.36 | mjordan | There's a cost with emitting information on every event. It isn't a cheap cost. |
| 17:20.50 | pjensen00 | Yeah, I was worried about the upstream acceptance. |
| 17:20.53 | pjensen00 | *am |
| 17:21.16 | mjordan | It'd be nice if the existing 'optionally, show this data' mechanism would work. |
| 17:21.25 | pjensen00 | But if I need to maintain my own minor patch, then it is what it is! I was just wondering if there was any value to the asterisk product by adding call IDs to the DialBegin |
| 17:21.51 | mjordan | It is somewhat specific, but that isn't to say that others wouldn't find it useful |
| 17:22.06 | mjordan | Specific - both in channel driver and in the particular piece of data. |
| 17:22.26 | pjensen00 | Do you know if the information is available on DialBegin or would it be just after that event? |
| 17:22.33 | pjensen00 | I'm not 100% sure on order of events there |
| 17:22.33 | mjordan | I'd prefer to find a more generic solution in this case. One thing that would, for example, solve it would be to have specific channel variables set by the channel drivers themselves |
| 17:22.48 | pjensen00 | Mmm yes. |
| 17:22.51 | mjordan | I'll have to look as well |
| 17:22.57 | mjordan | will be right back |
| 17:24.13 | mjordan | DialBegin will get emitted after ast_call is called on the channel, which means sip_call (in chan_sip - making an assumption that's what you're using) will be called first |
| 17:24.22 | mjordan | so any channel variables set there could be emitted on the DialBegin event |
| 17:24.46 | pjensen00 | awesome! |
| 17:24.53 | pjensen00 | thanks for checking, much appreciated. |
| 17:24.56 | mjordan | np |
| 17:25.06 | mjordan | if you run into any issues on writing said patch, feel free to hop in #asterisk-dev |
| 17:25.24 | file | ponders |
| 17:25.34 | mjordan | glances at file |
| 17:25.47 | file | mjordan, semi-unrelated |
| 17:26.05 | file | my mind just went off on a tangent of stateless ARI applications |
| 17:26.12 | mjordan | file: well, yes |
| 17:26.29 | mjordan | hm. |
| 17:26.38 | mjordan | I just realized I was probably wrong on ye olde blog post comment |
| 17:26.47 | mjordan | PJSIP_HEADER should get the Call-ID header from an INVITE |
| 17:27.10 | mjordan | sighs |
| 17:27.13 | mjordan | I'll have to go reply |
| 17:27.36 | pjensen00 | mjordan: Do you think it makes more sense to emit a new VarSet event or add the call ID to the Dial event on the PJSIP channel driver? |
| 17:28.40 | mjordan | pjensen00: well, in PJSIP land, we're actually in better shape. |
| 17:29.01 | mjordan | pjensen00: you can actually use channelvars=PJSIP_HEADER(read,Call-ID) |
| 17:29.31 | mjordan | and it should populate... something. I actually haven't tested that out, but the invocation of resolving the channelvars values does accept dialplan functions |
| 17:29.50 | mjordan | and since PJSIP_HEADER can read headers out of inbound/outbound legs, that should extract it. Again, haven't tested that yet. |
| 17:30.12 | pjensen00 | Yeah, I've done something similar for my hangup callback. |
| 17:30.37 | pjensen00 | I just need to patch it so it does that before emitting the DialBegin command and I should be golden. |
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| 17:31.44 | mjordan | Well, the ChannelVars setting in manager.conf is a bit funky. It will always resolve that particular function call on every event that involves a channel |
| 17:32.20 | mjordan | although with pjsip, it may not work as nicely. |
| 17:32.21 | mjordan | Hm. |
| 17:32.43 | mjordan | chan_pjsip will enqueue the call on another thread, so you may get into a race condition where the DialBegin does go out just prior to the actual INVITE request |
| 17:33.00 | pjensen00 | Hrm. |
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| 17:52.17 | pjensen00 | Well, looks like I'll have to queue up the DialBegin if I don't have the Call-ID yet. I think I can make that work. |
| 17:52.37 | pjensen00 | In my side of the event processing of course. |
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| 18:32.52 | irreverant | hello |
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| 18:37.15 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:2f70:21b:63ff:fe31:8426) |
| 18:37.15 | gusto | so |
| 18:37.20 | gusto | WIMPy: hi |
| 18:37.27 | gusto | WIMPy: how are you today? |
| 18:38.20 | WIMPy | Not too bad. |
| 18:40.29 | gusto | why? |
| 18:40.32 | gusto | what is wrong? |
| 18:40.49 | WIMPy | Nothing so far. |
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| 18:41.44 | gusto | you do not look very confident |
| 18:41.57 | gusto | i am better, since my father left this flat |
| 18:42.04 | gusto | with his new girlfriend |
| 18:42.25 | WIMPy | The day isn't over, yet. |
| 18:43.08 | WIMPy | Although I don;t plan to do any PC stuff, so chances are there won;t be anything bad happening. |
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| 18:51.02 | gusto | as long as you do not touch anything it will continue to work, you mean |
| 18:51.22 | WIMPy | hopes so |
| 18:51.55 | gusto | the paranoid administrator is in fear of intruders from the outside world |
| 18:52.03 | gusto | the experienced administrator is in fear of himself |
| 18:52.29 | Nivex | gusto: well put sir |
| 18:52.51 | WIMPy | The worst intruders from outside come whithe the initial installation. |
| 18:53.28 | gusto | with commercial products like microsoft, maybe |
| 18:53.49 | WIMPy | Doesn't matter. |
| 18:54.53 | gusto | as long as something is open-source one can always take a look to be sure |
| 18:56.01 | WIMPy | Yous, you can. |
| 18:56.10 | WIMPy | Do you do it? |
| 18:56.30 | WIMPy | If not itsn't any more secure than any commercial product. |
| 18:56.49 | WIMPy | And even if you do there's a very good chance that you wouldn't notice. |
| 18:57.58 | WIMPy | And everyone should know that at the latest since al the heartbleed etc things. |
| 18:58.04 | gusto | not me, but a lot of ppl do, and i do trust the mass more than i trust some company which has info-politics and whatever reasons of conduct |
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| 18:58.54 | bewbs | hi there. i'm trying to decide if asterisk or freepbx is more suited for my application. i'm starting iwth zero knoweldge not only of the systems, but phone systems in general. |
| 18:59.02 | gusto | yes, but there are consequences to that, something that could not come out of commercial products so easily, openbsd started libressl |
| 18:59.23 | tm1000 | bewbs: freepbx uses asterisk, so asterisk is asterisk |
| 18:59.25 | gusto | asterisk is ok |
| 18:59.34 | WIMPy | bewbs: That question doesn't make as much sense as you might think. FreePBX is a configuration frontend for Asterisk. |
| 18:59.44 | bewbs | that makes sense |
| 18:59.52 | bewbs | so front end config |
| 19:00.08 | bewbs | like a query tool sitting on a database |
| 19:00.22 | WIMPy | And it will take over all the configuration. |
| 19:00.33 | [TK]D-Fender | like webmin.. but only for Asterisk |
| 19:00.43 | bewbs | so is it an either/or or a first/then type situation |
| 19:00.50 | [TK]D-Fender | And isn't something you get to play around with manually so much once you hand over thw reigns |
| 19:02.15 | gusto | i do not trust frontends |
| 19:02.34 | gusto | it is hard enough to get asterisk doing what i want through .conf files |
| 19:02.51 | bewbs | so information found on asterisk site would also apply to freepbx, and vice versa |
| 19:02.54 | bewbs | right? |
| 19:03.18 | WIMPy | Not exactely. |
| 19:03.30 | WIMPy | The configuration is very much hidden away from you. |
| 19:03.37 | bewbs | asterisk would apply to pbx, but not necessarily vice versa |
| 19:03.46 | WIMPy | And the list of available featues might not be the same, either. |
| 19:03.52 | bewbs | what is htere to configure other than the devices on the network? |
| 19:04.03 | WIMPy | doesn't understand the pbx comment. |
| 19:04.18 | WIMPy | Everything. |
| 19:04.46 | WIMPy | You have to implement almost every of the standard pbx features yourself. |
| 19:04.47 | bewbs | does the system require devices be on their own vlan, or can they be on a common lan |
| 19:05.00 | tm1000 | bewbs: either way |
| 19:05.01 | WIMPy | That's what FreePBX would have done for you (hopefully). |
| 19:05.01 | [TK]D-Fender | bewbs: You define ALL of that. |
| 19:05.18 | bewbs | you mean caller ID, phone association id's, etc |
| 19:05.23 | bewbs | broadcast |
| 19:05.34 | WIMPy | I wouldn't recomment sharing the same network for VOIP and other uses. |
| 19:05.42 | gusto | ~book |
| 19:05.43 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:06.17 | bewbs | this sounds exciting |
| 19:06.22 | WIMPy | Actually I wouldn't recommend COIP at all. |
| 19:06.27 | WIMPy | VOIP |
| 19:06.33 | bewbs | what's the atlernative? |
| 19:06.53 | WIMPy | The good old hardware we used so far. |
| 19:07.07 | bewbs | what are the disadvantages? |
| 19:07.32 | WIMPy | cost, cost and functionality. |
| 19:07.38 | bewbs | hmm |
| 19:08.08 | bewbs | so i can't just connect some ip phones to my switch, fire up a vm with said software, and have it go? |
| 19:08.22 | WIMPy | Yes you could. |
| 19:08.49 | WIMPy | But be prepared for some robotic voices if someone transfers files while you're on the phone. |
| 19:09.02 | bewbs | i've gotta do this discretely before my wife figures out what i'm doing |
| 19:09.24 | bewbs | so create a vlan on the switch and have phones on that? |
| 19:09.51 | [TK]D-Fender | bewbs: WIMPy has some very stringent expectations of PBX's that does not seem to align with the general population. For just about everyone else.. it's fine. feel free to ask him the particulars and see if his issues with it would issues for you as well |
| 19:09.53 | WIMPy | And configure the switch to prioritize that vlan. |
| 19:10.15 | bewbs | someone's always transfering files, vm cluster accesses vhd's on shared storage |
| 19:10.25 | bewbs | i just like to play with new shit |
| 19:10.27 | bewbs | it fascinates me |
| 19:10.33 | bewbs | i'm a tinkerer |
| 19:10.41 | WIMPy | knows that issue |
| 19:10.53 | bewbs | the idea of a half dozen phones plus a front door intercom is awesome |
| 19:11.03 | bewbs | pick up phone at my desk, hit 6, get kid #2 |
| 19:11.05 | gusto | i hear old hardware? not a good idea |
| 19:11.07 | bewbs | "you done your homework?" |
| 19:11.21 | bewbs | define "old" |
| 19:11.33 | gusto | i have my asterisk here running on a beagle bone black, that is much faster than raspberrypi and with archlinux its working very good |
| 19:11.42 | WIMPy | Stoff from the last two decades. |
| 19:11.54 | WIMPy | Stuff.. |
| 19:12.41 | bewbs | i have 5 pe1950's clustered vm servers, a pe2950 with 6x240gb ssd's for shared storage, 2 nortel baystack 5520's with poe, and a supermicro with 8x4tb hdd's for my nas |
| 19:12.56 | bewbs | i don't know if htat's to old or not |
| 19:13.39 | WIMPy | Doesn't matter. People use Raspberry Pis just fine. I was talking about phones. |
| 19:13.54 | bewbs | http://i.imgur.com/4UnF8jS.jpg |
| 19:14.04 | bewbs | i was reserving switch 1 for poe devices |
| 19:14.10 | bewbs | cameras and phones |
| 19:15.01 | WIMPy | Big switches are evil. They suck away all your money. |
| 19:15.10 | bewbs | how? |
| 19:15.30 | WIMPy | Power consumption. |
| 19:15.53 | bewbs | with everything on the rack, i think the switches are hte least of my issues |
| 19:15.54 | WIMPy | Also a big issue with IP phones themselves, as well as them needing switch ports. |
| 19:16.34 | WIMPy | I didn't think they were that bad until a friend convinced me to take a measurement. |
| 19:16.46 | WIMPy | I retired it the next day. |
| 19:16.59 | bewbs | each of hte 1950's have dual 670w psu's |
| 19:17.07 | bewbs | the rack needs a dedicated 20a breaker |
| 19:17.33 | bewbs | but it's not pushing power to phones unless they're in use |
| 19:17.56 | WIMPy | They already use a lot of power when idle. |
| 19:19.17 | bewbs | i put them in production 8mo ago |
| 19:19.32 | bewbs | powerbill has not gone over 500/mo |
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| 19:19.56 | WIMPy | 500 what? |
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| 19:20.39 | bewbs | dollars |
| 19:21.30 | WIMPy | I'd cerrtainly not be prepared to pay hundreds per month for electricity. |
| 19:21.49 | bewbs | i've been doing it for years |
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| 19:22.31 | bewbs | between the mini data-center, mini-aquarium, large AC units, pool pump, plus kids htat can't turn off a light to save hteir lives |
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| 19:30.21 | SpaceInvaders | Hi, can Asterisk handle Ringmaster? Or is that an ata question? Or both? Or something else (I'm no expert)!!?? Thanks! |
| 19:30.51 | WIMPy | What is "ringmaster"? |
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| 19:39.44 | mjordan | WIMPy: I don't know, but for some reason I had flashbacks to GhostBusters (Are you the keymaster?) |
| 19:40.37 | WIMPy | can't remember the story. That's too been long ago. |
| 19:40.49 | WIMPy | err, been too |
| 19:41.58 | _Corey_ | 'ringmaster' would be a good IRC handle |
| 19:44.07 | Qwell | _Corey_: -NickServ- Registered : Nov 17 21:26:34 2004 (9 years, 30 weeks, 3 days, 22:17:13 ago) |
| 19:45.19 | _Corey_ | sounds about right ;) |
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| 19:59.18 | P-NuT | Hi there, is there any way to do video conferencing with asterisk? |
| 19:59.48 | [TK]D-Fender | P-NuT: * will do "follow the speaker" only, and all of the endpoints have to be using the same codecs |
| 19:59.56 | WIMPy | Depends on your definition for "video conferencing" |
| 20:03.11 | P-NuT | What I mean is a 3 way video conference not unlike you can do with oovoo or skype. |
| 20:03.47 | mjordan | as long as you don't want little picture boxes ala Google Hangouts, then yes |
| 20:03.53 | mjordan | use ConfBridge |
| 20:04.10 | SpaceInvaders | ringmaster allows several phone numbers to be used on one line. Each phone number is given a unique ring by the phone company |
| 20:04.20 | [TK]D-Fender | mjordan: Still doesn't MUX vide, does it? |
| 20:04.36 | [TK]D-Fender | mjordan: Last I heard it is still "follow the speaker" |
| 20:04.47 | mjordan | nope! It still does single presenter. You have a few more options - locking speaker, etc. - but it is just follow the speaker |
| 20:05.01 | WIMPy | SpaceInvaders: That's usually known as Destinctive Ring and has to be done by whatever you physically connected to the line. |
| 20:05.06 | mjordan | If what you want are three picture boxes, then yeah, you're out of luck |
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| 20:05.40 | P-NuT | [TK}D-Fender: I am talking about having a system mux video. |
| 20:05.47 | [TK]D-Fender | P-NuT: Not happening |
| 20:05.54 | SpaceInvaders | so if I connect an ata that supports distinctive ring then Asterisk can ring a specific extension? |
| 20:05.58 | WIMPy | mjordan: What was the role of the keymaster? And are you the gatekeeper or whatever the other guy was called? |
| 20:06.14 | mjordan | WIMPy: yeah, that was what I was alluding to. They opened the gateway for Zuul |
| 20:06.32 | WIMPy | SpaceInvaders: Whatever that ATA can be configured for. |
| 20:06.37 | mjordan | Which, funny enough, is the name of an OpenStack project... |
| 20:06.49 | mjordan | And yes, P-NuT - Asterisk is not a video mux. |
| 20:07.12 | mjordan | It provides some decent *free* video conferencing capabilities. If you want to get anything better than that, you'll have to go for a commercial solution |
| 20:07.12 | SpaceInvaders | so if the ata supports "ringtone1" and "ringtone2" does it pass that to Asterisk? And how? |
| 20:07.22 | SpaceInvaders | when I say how I mean how does asterisk receive it? |
| 20:07.34 | WIMPy | SpaceInvaders: See its manual. |
| 20:08.37 | SpaceInvaders | I don't have the ata yet. Do all atas support distinctive ring? I was told the obi110 was a good one to get and that's what I was thinking of ordering |
| 20:09.03 | WIMPy | Certainly not. |
| 20:09.22 | WIMPy | And if they do, they most probably do it in different ways. |
| 20:09.39 | WIMPy | So check the manuals of the ones you think of getting. |
| 20:09.58 | WIMPy | Or better still, forget about the whole thing and go digital. |
| 20:10.07 | SpaceInvaders | I should have thought of looking for a soft copy of the manual |
| 20:10.22 | SpaceInvaders | Ya I've thought about going 100% digital |
| 20:10.25 | SpaceInvaders | but |
| 20:10.29 | SpaceInvaders | I think my wife would explode |
| 20:10.29 | WIMPy | Do it |
| 20:10.44 | SpaceInvaders | oh now your just looking for fireworks! |
| 20:11.14 | SpaceInvaders | should I send you my dna profile so you can verify chunks if they hit as far as you |
| 20:11.22 | SpaceInvaders | are you on North America? :D |
| 20:11.46 | WIMPy | Does your wife have digiphobia or something? |
| 20:11.52 | SpaceInvaders | and thank you for your help :) |
| 20:11.58 | SpaceInvaders | oh no she just hates change |
| 20:11.58 | WIMPy | Nope. Far enough away. |
| 20:12.25 | SpaceInvaders | once I do something and show it to her thats my best chance |
| 20:12.39 | WIMPy | Then do it. |
| 20:12.51 | SpaceInvaders | as long as it's something I can "show" her rather than telling her she's stuck with it |
| 20:13.22 | WIMPy | DR on a single POTS line doesn't really make much sense anyway, does it? |
| 20:15.02 | SpaceInvaders | it could be used as a white-list |
| 20:15.35 | SpaceInvaders | and the white-list line could ring all extensions or specific ones |
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| 20:16.01 | SpaceInvaders | especially home# vs home-office# |
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| 20:21.49 | practichem | Hey all, anyone have a workaround so I don't have to press "1" for new messages in voicemail? |
| 20:21.54 | WIMPy | Make your own. |
| 20:21.57 | practichem | Hoping something was out there first! |
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| 20:33.17 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 20:41.29 | Nivex | ok, what am I doing wrong? Incoming calls from IPKall show the CID with the IPKall number not the actual caller. |
| 20:42.23 | Nivex | incoming calls through Google Voice that is |
| 20:42.33 | Nivex | dialling straight to IPKall shows the correct CID |
| 20:44.59 | TazzNZ | Nivex: are you sure they are sending you the CID |
| 20:45.04 | TazzNZ | oh |
| 20:45.37 | TazzNZ | so calls coming in via Google Voice shows the IPKall number ? |
| 20:45.40 | Nivex | yes |
| 20:45.49 | TazzNZ | as the number that is dialing you ? |
| 20:46.04 | TazzNZ | you will have to paste bin your extentions.conf and sip.conf |
| 20:50.44 | Nivex | kjotte@mystic:/tmp$ pastebinit /tmp/sip.conf-sanitized |
| 20:50.44 | Nivex | http://paste.ubuntu.com/7660320/ |
| 20:50.44 | Nivex | kjotte@mystic:/tmp$ pastebinit /tmp/extensions.conf |
| 20:50.44 | Nivex | http://paste.ubuntu.com/7660321/ |
| 20:51.21 | Nivex | sanitized only to remove secrets. everything else is untouched |
| 20:54.09 | Nivex | (random aside) I really need to remove the GTalk outbound stuff |
| 20:59.57 | TazzNZ | geez - you really should remove all the default stuff too Nivex |
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| 21:03.16 | TazzNZ | same => n,Set(CALLERID(num)=${SHIFT(CALLERID(name),@)}) <-- why are you doing this ? |
| 21:03.24 | TazzNZ | in [google-in] |
| 21:03.58 | Nivex | because the CID coming in over chan_motif was pretty screwy |
| 21:04.14 | Nivex | that context shouldn't be hit anymore |
| 21:04.49 | TazzNZ | so where is your google inbound context then ? |
| 21:05.11 | Nivex | Google Voice forwards to IPKall, so it's coming in through the ipkall context |
| 21:05.22 | TazzNZ | right |
| 21:05.47 | TazzNZ | so, enable sip debugging and have a look at the headers that ipkall sends you |
| 21:05.58 | TazzNZ | sip set debug peer ipkall |
| 21:08.06 | Nivex | oh, hmm... it's not my ipkall number. it's from the same area code though. wonder if that's the GV exchange point there. still worth a look |
| 21:08.40 | TazzNZ | Nivex | Google Voice forwards to IPKall, so it's coming in through the ipkall context |
| 21:08.44 | TazzNZ | that is what you said |
| 21:08.50 | TazzNZ | so the call will come from IPKall |
| 21:09.04 | Nivex | yeah, but GV should be sending through the CID of the caller |
| 21:09.12 | Nivex | that's what I've got in settings |
| 21:09.42 | Nivex | ... anyway, I can't call my own GV number from my cell. I'll have to wait until later when I can borrow a friend's phone. |
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| 21:10.34 | TazzNZ | if you are fowarding the call from GV, GV might set your outbound number to your inbound number |
| 21:10.40 | TazzNZ | it kinda makes sense |
| 21:10.50 | TazzNZ | so that if you call, as example, me from GV |
| 21:11.02 | TazzNZ | I will see your inbound number, and I can call you back on that |
| 21:11.36 | Nivex | that's what I first thought, but I just double checked the numbers. my ipkall number is 206-279-5226 but the CID I get on incoming calls is 206-682-0185 |
| 21:12.26 | TazzNZ | so when someone calls your GV number, that is forwarded to IPKall, right ? |
| 21:13.26 | Nivex | ah, I think I found a possibility: http://www.dslreports.com/forum/r29037554-Google-Voice-sending-fake-CID-to-IPKall |
| 21:14.34 | Nivex | I thought it was something I had done, but it looks like it's something GV has done recently |
| 21:18.42 | Nivex | as the commenters say, I'm getting what I pay for |
| 21:18.48 | Nivex | Thanks for taking the time TazzNZ |
| 21:19.25 | TazzNZ | haha - np Nivex |
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| 23:24.06 | SpaceInvaders | in the guide where it says |
| 23:24.08 | SpaceInvaders | exten => 100,1,Dial(SIP/0000FFFF0001) |
| 23:24.17 | SpaceInvaders | exten means it's identifying extension 100, right? |
| 23:24.22 | SpaceInvaders | and what does the ,1 mean? |
| 23:24.27 | SpaceInvaders | 1 line? |
| 23:25.14 | pjensen00 | sequence |
| 23:25.34 | pjensen00 | exten => 100,2,Somethingelse |
| 23:25.43 | pjensen00 | would be next |
| 23:26.01 | SpaceInvaders | can you explain with a scenario? |
| 23:26.13 | SpaceInvaders | because I'm not getting it other than to understand "sequence" |
| 23:26.24 | SpaceInvaders | why would 100 have another "somethingelse" |
| 23:26.45 | pjensen00 | uh, sure. One second. |
| 23:26.48 | SpaceInvaders | thanks :) |
| 23:27.08 | SpaceInvaders | and I'm thinking... would it be if you wanted to have 100 ring a series of locations? maybe in order? |
| 23:27.09 | pjensen00 | ;exten => _X.,1,Answer(); |
| 23:27.10 | pjensen00 | exten => _X.,n,Wait(1) |
| 23:27.14 | pjensen00 | er |
| 23:27.17 | pjensen00 | exten => _X.,2,Wait(1) |
| 23:27.28 | pjensen00 | this is just a way of saying "Do 1 first, 2 second" |
| 23:28.00 | SpaceInvaders | ok I'm starting to understand and I'm starting to understand there's a lot I don't know ! ! :) |
| 23:28.11 | pjensen00 | There's more you can do |
| 23:28.16 | SpaceInvaders | and I do get it--"somethingelse" is if you want something else to happen after |
| 23:28.29 | SpaceInvaders | hehe ya like you said "there's more you can do" |
| 23:28.33 | pjensen00 | yeah, just some other dialplan command |
| 23:28.36 | SpaceInvaders | thanks for clarifying |
| 23:29.04 | pjensen00 | no problem |
| 23:29.23 | SpaceInvaders | if I set up the hello-world part will it work just by me adding the 3 lines to the extensions.conf from my default install on fedora using yum? |
| 23:29.40 | [TK]D-Fender | "default" doesn't mean anything really. |
| 23:30.02 | [TK]D-Fender | And we don't necessarily know what Fedor's packages come with in them as-is |
| 23:30.16 | SpaceInvaders | OK I'll just add it and see if it works, does nothing, or explodes :) |
| 23:30.24 | [TK]D-Fender | Also you need to properly understand the concept of contexts in the dialplan |
| 23:30.38 | SpaceInvaders | ya I'm still working on understanding exactly that |
| 23:30.43 | [TK]D-Fender | Because shoving 3 lines is "just anywhere" will not work. |
| 23:30.48 | [TK]D-Fender | They need to be in the RIGHT place |
| 23:31.01 | SpaceInvaders | oh that part I understand but not because I understand context |
| 23:31.11 | SpaceInvaders | just experience with cobol |
| 23:31.13 | [TK]D-Fender | Just like pouring gasoline in your windshield washer resevoir may be "putting gas in your car", but you aren't going to get very far |
| 23:31.18 | SpaceInvaders | if it's not in the right section, it won't work |
| 23:31.23 | [TK]D-Fender | correct |
| 23:31.28 | SpaceInvaders | ohh I like your example better :D |
| 23:32.11 | SpaceInvaders | I have a feeling to really understand contexts I'll have to play with it some |
| 23:32.33 | SpaceInvaders | but I'm having a blast learning about it and really apprecate the help found here. Thank you very much. |
| 23:32.59 | [TK]D-Fender | Won't be a choice because you'll need them to separate the stuff coming in from phone-type devices (users0, from processing calls in from the outside world |
| 23:33.21 | [TK]D-Fender | You wouldn't want some random call in to accidently be allowed to go right back out. |
| 23:33.53 | SpaceInvaders | Exactly. Right now I only have confidence because I have some expertise with firewalls |
| 23:34.09 | SpaceInvaders | and my Asterisk is well firewalled :) |
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